atcomsystems.ca/forum Forums General General Telephone & Systems Information disconnect supervision

I have a T3 from Qwest (quest | whatever) going into a DS-3 Mux. The T1s from that mux are going into a CAC channel bank with 24 FXS ports. Those FXS ports are connected to a Systembase VoIP gateway.

Here is the problem:
The provider (qwest) is only capable of supplying an AB-Bit to signal the end of call. The VoIP gateway cannot recognize these AB-Bits. It is made for the Taiwan market where all the signaling is analog. I need to translate an AB-Bit into an analog busy tone (or some other analog disconnect signal).

Is there anything out there that will generate this tone when it receives an AB-Bit?

Can you set the silence (no talk) threshhold smaller?
This is a common problem with alot of switchs that need disconnect supervision.

Can your switch except a T1? a T3(7 T1's)is kind of large for alot of applications.

The channel bank with the FXS(analog) ports is the problem. They are receiving the AB sequence. What brand or channel bank are you using? Most can be programmed to reconize the disconnect bits and provide a battery reversal or a timmed no battery condition.

The channel bank is a carrier access. Also, what were you saying about battery reversal? I think the VoIP gateway is very specific in what it needs to disconnect. I'll mention that though...thanks.

The CAC channel bank is capable of sending 'open loop disconnect'. Basically this is the deal: The channel signaling between the carrier (in this case Qwest) and the channel bank should be set as E&M wink start. Between the channel bank and the analog CPE equipment will be set as standard POTS type loop start lines. When the carrier side of the line disconnects first the carriers A-B bits will go from 'high' to 'low'. This will result in the CAC channel bank to send a 2 second open loop to the associated analog POTS line. Your analog interface to the gateway should recognize this as the disconnect supervision.

There are two cards in the channel bank each has a set of dip switches labeled:
switch option A
switch option B
Switch option C

Set the Switch option A to 'on' on each card (one card is the first 12 the other 13-24). Remember you will need the carrier to change to E&M wink start for this to work. As it is now if the A option is set to off you will be using loop start with no 'open loop disconnect'.

Here is the summary from the CAC manual:


E & M Wink-Start to Loop-Start Conversion (AT&T Megacom, Flexpath, etc.) with Calling Party Disconnect
Access Bank FXS interfaces provide Plain Old Telephone (POTS) Loop-Start interfaces to a phone system
or regular telephones. E&M wink-start signaling and per-channel ringback tones are provided to the T1
line. Calling Party Disconnect is a two second Tip Open condition (loop current turns off) at the FXS when the
network releases seizure (the far-end call hangs up). Battery feed from the FXS returns to normal idle
(Tip grounded) after the two second disconnect. This feature is similar to local telephone service Calling
Party Disconnect (CPD), provided by some central office equipment for answering machines and PBXs
which are equipped to detect loss of loop current on their line interfaces. This loss of loop current informs
the equipment to hang up.
Switch Option A ON
Switch Option B Off
Switch Option C Off

Having said all that it seems easier to bring the t1 directly in to the gateway, not using a channel bank break out at all.

It is much easier to bring the T1 directly into the gateway...I agree. Unfortunately, the gateway is analog, not digital, and it only accepts FXS connections to its FXO ports. Also, we have exhausted all tinkerings with the channel bank. The issue is the Gateway. The only way it will disconnect the call is with a disconnect tone or busy tone in the 500-700Hz range. This is a quote from the Gateway manufacturer.

I am working with the people who make the Linux based PBX...Asterisk. They already told me that it can produce these tones. I now need to determine configuration and capacity.

Any other suggestions would be welcomed. The only option is to produce an tone on the 500-700Hz range that signals the end of call. If there is hardware out there that can translate an AB-Bit into a busy tone, please let me know. Also, I would rather make the translation at the T1 level, not the DS0 level.


I guess this is what happens when you save money on equipment and billing software by getting off brand equipment.